a. Field of the Invention
The present invention relates generally to passing voice communications over a data communications network such as an asynchronous communications network or a synchronous communications network.                b. Background Information        
Almost all customers of data traffic today have additional, separate links to carry voice. This is inefficient for the customer and the communications provider. Many are seeking techniques that place Ds0 channels in data packets for transmission over a data link, so that they can remove their voice links.
A communications network serves to transport information among a number of locations. The information is usually presented to the network in the form of time-domain electrical signals and can represent any combination of voice, video, or computer data. A typical communications network consists of various physical sites called “nodes,” interconnected by conduits called “links.” Each link carries information from one site to another site. End user sites may contain data terminating equipment (DTE) for combining, separating, and transforming data with or without voice. Network provider sites usually include either edge switches, with user network interfaces (UNI), or backbone switches, which only connect to other backbone switches and edge switches and do not contain UNI.
Voice information is carried via a Ds0 (or voice) channel that is a 64 kilobits per second (64 Kbps) channel and also the worldwide standard for digitizing voice conversation. The channel throughput is 64 Kbps because a digital data stream can adequately represent an analog voice signal if sampled at a rate of 8000 samples per second. If each voice sample is digitized using 8 bits, this results in a digital data stream of 64 Kbps. Since Ds0 is a synchronous TDM link, once a channel connection has been setup between two users, that channel is dedicated until the connection is torn (or brought) down, and cannot be used by anything or anybody else even if there is silence in the line.
Data currently is transmitted between nodes either as synchronous or asynchronous. In a synchronous network using Synchronous Transfer Mode (STM), each timeslot is assigned a certain time when it is to arrive at each node. The time when the timeslot arrives determines where the timeslot goes. Thus, the individual timeslots do not need to have routing information within them.
Asynchronous Transfer Mode (ATM), Frame Relay (FR), and Internet Protocol (IP), collectively called data, are considered asynchronous because each node in the network does not know until after a data packet arrives where it is intended to go. The arrival of a particular data packet at a node, on the other hand, is not guaranteed to occur at a particular point in time. Only by analyzing the routing information in the header can the data switch know where to route the data packet.
Asynchronous Transfer Mode is designed to be carried over the emerging fiber optical network, called the Synchronous Optical NETwork (SONET), although it can be carried over almost any communications link. The basic unit of ATM is a data packet called the ATM cell. Each cell contains two parts, a header, which contains routing information, and a payload, which contains the data to be transported from one end node to another. The ATM cell is always the same size.
Frame Relay and Internet Protocol are two other asynchronous types of communications protocols. Each is similar to ATM in Fat they also consist of a data packet. However, they differ from ATM in that their packet size can vary from packet to packet, and both can be considerably larger than ATM. This allows them to make more efficient use of the bandwidth of the communications media they travel over, but it makes receiving them more difficult in that packet size must be calculated for each packet. Both the FR protocol and IP may be used in point to point connections, but IP may also be used when multiple ports are connected to a single transmission medium.
Data can consume as much or as little as is needed for carrying actual traffic, because data does not reserve a fixed amount of bandwidth per link. While voice will never overload, or oversubscribe, the capacity of its links, there are mechanisms in place to handle data overloads when more is available than a physical link can carry. It is these mechanisms that allow data network designers to specify more data demand than capacity to carry, which is a process called statistical multiplexing.
Statistical multiplexing is the concept of giving multiple customers, in sum total, more bandwidth through a physical connection than it can carry. This is also known as over-subscribing. Studies have shown that customers will not always use all of the bandwidth their carrier has set aside for them. It is during this period of non-use by a customer that spare bandwidth is available for the over-subscription. If sufficient numbers of customers are placed on a single physical connection then large quantities of spare bandwidth can be realized.
When traffic is isolated among two or more physical connections, less statistical multiplexing can occur, as customers on one connection cannot use spare bandwidth on another. By joining all customers into a single, large connection, better statistical multiplexing occurs and the carrier is able to sell more bandwidth on one high-speed physical connection than on several smaller connections whose sum is equal to the one high-speed connection.
There are different ways of handling overloads in the data network. In ATM, the network is designed with large buffers, which absorb the excess traffic, queuing it up until capacity is available to place the traffic through the network. The traffic that is delivered out of its buffers first is determined by the quality of service (QOS) the customer has paid the carrier to provide. Higher QOS traffic is removed from its buffers before lower QOS. This is important for real time applications such as voice or interactive TV services, which must get through the network with a minimum amount of delay.
In those instances where so much excess traffic is delivered that the network cannot queue it up in buffers, the lower QOS traffic is deleted, or dropped, from the buffers to make room for higher QOS traffic to be queued up. Ideally, customer end-to-end protocols will detect this loss of traffic and will re-transmit the lost information.
An emerging standard in the IP network uses a different approach to handling overloads. In IP, there is no QOS as in ATM. Once a data packet is injected into the IP network, it will be given equal priority with all other traffic and delivered to its destination with a minimum of delay.
In an IP network, the traffic density in a link is closely monitored. As it begins to approach the link capacity, the IP data switch send congestion notices back towards the data sources telling them to slow down the amount of data they send. Each notice will, for a limited length of time, force the data source to restrict what it sends out. As link traffic gets closer and closer to link capacity, more of these messages are sent backwards. When an IP switch receives congestion notices and reduces the rate of transmission, it may experience congestion as well and will send congestion notices back to its sources.
Eventually, the notices reach the traffic origins, customers. The customer equipment must then cut back on what is sent into the network, and must decide which traffic it puts out has the highest priority so that it goes out while the lower priority traffic has to wait. Thus, the IP network passes the job of determining traffic priority to the customer. If a customer has a great deal of high priority traffic, it may pay a premium to not receive as many congestion notices when congestion hits the network as another customer may pay, so that it will get more guaranteed traffic during congestion.
The IP data switches also usually maintain small buffers, but these are designed exclusively to handle the small, temporary overloads that occur until the congestion notices are responded to and reduced traffic flows into the switch.
These two different means of determining traffic priority are given as an example only. Whatever the mechanism, voice will usually be given a higher priority than data. By ensuring that the voice traffic does not physically exceed the capacity of the network links, the network systems engineering team can ensure all voice gets through, squeezing out the needs of data traffic. This allows the physical links to stay at or close to capacity even as the demands of Ds0 change. This spreads the cost of the links out over more traffic, reducing the cost per data packet and thus making the network more efficient than dedicated links carrying voice can be.
Each of ATM, FR, and IP has certain benefits and certain disadvantages. By utilizing these protocols in areas where their benefits can be utilized to the maximum efficiency while minimizing the losses incurred from their disadvantages, a more efficient network is realized.
Because of its fixed size packet, ATM is more attractive on the higher speed links where it is considerably less expensive to design hardware to receive a fixed size packet than a variable sized packet. On the lower speed, but higher per-bit cost links, FR and IP are more attractive because of their line utilization efficiency. And at these speeds the cost difference between a port that can receive variable sized packets versus one that only has to receive fixed size packets is usually more than offset by the fact that there are no segmentation and reassembly functions that have to be performed. Segmentation and reassembly is needed when a variable sized message is placed in multiple data packets, which is necessary with ATM.
Improvements in the state of the art of design technology are making the segmentation and reassembly functions less expensive. On the other hands, similar improvements are making it easier to design IP and FR receivers that can operate at SONET rates. What will likely happen is that the industry will see ATM, which is maturer than IP, dominate the high-speed market for the next 5 to 10 years. After that time period, IP, which has a simpler and less expensive congestion management scheme than ATM, will become the dominant mechanism in high-speed traffic.
On a bit per bit basis, it is significantly less expensive to transmit data over fiber than using metallic-links by several factors of ten. The theoretical capacity of fiber is in excess of 20 tera bits per second (20 million million bits per second). Current standards at 10 thousand million bits per second (gigabits per second, or gbps), and will soon increase to 40 thousand million bits per second. Furthermore, technology is also improving on the ability of a single fiber to carry numerous wavelengths, or colors. Each wavelength can carry 10 gbps independently of what the other wavelengths in the fiber are doing.
On the other hand, metallic links that can span long distances and are reasonable to manufacture have long ago reached their theoretical limits of roughly under 500 million bits per second. They are much bulkier than fiber optic links. The metallic link is also susceptible to rust and corrosion, whereas the fiber is relatively chemically inert.
A T1 link, which is an example of a metallic link, transmits one T1 frame 8000 times per second (or one frame every 125 μs). Each T1 frame contains a T1 payload with 24 Ds0 timeslots, one for each Ds0 channel with 8 bits in each timeslot. Each T1 frame also has a T1 frame bit that identifies the start of the T1 frame, so that a T1 frame has a total size of 193 bits. This results in a data stream of 1.544 Mbps (8000 frames/sec 193 bits/frame).
Repeaters, which re-amplify the signal, are needed to prevent signal attenuation (loss of signal strength as a signal travels down a link) on either type of link. Metallic links attenuate the signals more than do fiber links, so more repeaters for metallic links are needed than for fiber links for a given distance. For instance, a T1 link can span a maximum of just over one mile (6000 feet) before a repeater is needed; for T3, the range is under 1400 feet. It is not unusual for fiber optic links to span 50 to 100 miles between repeaters. Fiber also costs less per foot physically than metallic links do, and the connectors at each end of a fiber link are similar in price to the connectors of a metallic link. Given the longer span between repeaters, this translates into fewer connectors, and hence lower costs, for fiber.
While metallic interfaces on port cards and repeaters are less expensive than fiber interfaces, the cost difference does not justify the reduced number of repeaters in a fiber network, nor does it justify the more expensive cabling needs even inside a switching facility. Further, the limited range of T3 metallic links has impacted the designs of several switching facilities, whereas the range of fiber links does not factor into their design.
c. Related Art
There are a variety of existing algorithms for removing silent voice channels in data switches. The primary approaches to handling silent and/or unused channels involve the use of algorithms at each node.
These approaches require a digital signal processor to analyze Ds0 channels to determine if they are silent or unused. The digital signal processor can only handle a fraction of the Ds0 channels that pass through a port card, requiring multiple digital signal processors to handle all of the traffic on the port card. The digital signal processor also induces additional delay in the passage of traffic through the port card so it can analyze the energy content of the Ds0 signals to determine if it is silent or not.
However, not all algorithms may be recognized or supported by every data switch that contains a Ds0 switching function within it. Even when an algorithm is supported, the algorithm will usually require a large amount of processing capacity from a digital signal processor. Current technology can handle several hundred voice channels per digital signal processor. A typical port card with a 2.5 gbps capacity can carry in excess of 16,000 Ds0 channels. Assuming 250 voice channels per digital signal processor, it would take 64 digital signal processors to examine and process silence suppression for each voice channel. It is impractical to place this many digital signal processors on a port card and will remain so for many years. Even with improvements in processing speeds and capacity of digital signal processors on port cards, port speeds will increase by a factor of about four in the next two years, and a factor of about sixteen a few years after that. There will still need to be multiple digital signal processors on a port card, which is still inefficient and not practical because of space considerations, and costs.
Another problem with existing silence suppression techniques is that each technique requires a large sample of voice frames in order to determine if the voice signal is silent or not at each switch in the communications network. This large sample slows down the transition of the voice channel from one end of the network to another, increasing latency, or end to end delay, to where it is unacceptable.
Notwithstanding the usefulness of the above-described voice switching matrices and port cards, a need still exists for a method or a digital signal processor to remove unused or silent channels from a data transmission to lessen the traffic through a data network or communications network.